def __init__(self, filename):
self.name = filename
#load/set wav file
if len(sys.argv) < 2:
print("Usage: %s wavefile" % os.path.basename(sys.argv[0]))
print(" Using an example wav file...")
dirname = os.path.dirname(os.path.realpath(__file__))
fname = os.path.join(dirname, filename)
else:
fname = sys.argv[1]
wavefp = wave.open(fname)
channels = wavefp.getnchannels()
bitrate = wavefp.getsampwidth() * 8
samplerate = wavefp.getframerate()
wavbuf = wavefp.readframes(wavefp.getnframes())
self.duration = (len(wavbuf) / float(samplerate)) / 2
self.length = len(wavbuf)
formatmap = {
(1, 8): al.AL_FORMAT_MONO8,
(2, 8): al.AL_FORMAT_STEREO8,
(1, 16): al.AL_FORMAT_MONO16,
(2, 16): al.AL_FORMAT_STEREO16,
}
alformat = formatmap[(channels, bitrate)]
self.buf = al.ALuint(0)
al.alGenBuffers(1, self.buf)
#allocate buffer space to: buffer, format, data, len(data), and samplerate
al.alBufferData(self.buf, alformat, wavbuf, len(wavbuf), samplerate)
def __init__(self, audio_format):
super(OpenALAudioPlayer, self).__init__(audio_format)
try:
self._al_format = format_map[(audio_format.channels,
audio_format.sample_size)]
except KeyError:
raise OpenALException('Unsupported audio format.')
self._al_source = al.ALuint()
al.alGenSources(1, self._al_source)
# Seconds of audio currently queued not processed (estimate)
self._buffered_time = 0.0
# Seconds of audio into current (head) buffer
self._current_buffer_time = 0.0
# List of (timestamp, duration) corresponding to currently queued AL
# buffers
self._timestamps = []
# OpenAL 1.0 timestamp interpolation
self._timestamp_system_time = 0.0
# Desired play state (True even if stopped due to underrun)
self._playing = False
# Timestamp when paused
self._pause_timestamp = 0.0
self._eos_count = 0
def write(self, audio_data):
buffer = al.ALuint()
al.alGenBuffers(1, buffer)
al.alBufferData(buffer, self._al_format, audio_data.data,
audio_data.length, self.audio_format.sample_rate)
al.alSourceQueueBuffers(self._al_source, 1, ctypes.byref(buffer))
self._buffered_time += audio_data.duration
self._timestamps.append((audio_data.timestamp, audio_data.duration))
audio_data.consume(audio_data.length, self.audio_format)
def __init__(self):
self.source = al.ALuint(0)
al.alGenSources(1, self.source)
al.alSourcef(self.source, al.AL_ROLLOFF_FACTOR, 0)
al.alSourcei(self.source, al.AL_SOURCE_RELATIVE, 0)
self.state = al.ALint(0)
self._volume = 1.0
self._pitch = 1.0
self._position = [0, 0, 0]
self._rolloff = 1.0
self._loop = False
self.queue = []
def __init__(self):
#load source player
self.source = al.ALuint(0)
al.alGenSources(1, self.source)
#disable rolloff factor by default
al.alSourcef(self.source, al.AL_ROLLOFF_FACTOR, 0)
#disable source relative by default
al.alSourcei(self.source, al.AL_SOURCE_RELATIVE, 0)
#capture player state buffer
self.state = al.ALint(0)
#set internal variable tracking
self._volume = 1.0
self._pitch = 1.0
self._position = [0, 0, 0]
self._rolloff = 1.0
self._loop = False
self.queue = []
def __init__(self):
self.channels = 1
self.bitrate = 16
self.samplerate = 8000
self.wavbuf = None
self.alformat = al.AL_FORMAT_MONO16
self.length = None
## formatmap = {
## (1, 8) : al.AL_FORMAT_MONO8,
## (2, 8) : al.AL_FORMAT_STEREO8,
## (1, 16): al.AL_FORMAT_MONO16,
## (2, 16) : al.AL_FORMAT_STEREO16,
## }
## alformat = formatmap[(channels, bitrate)]
self.buf = al.ALuint(0)
al.alGenBuffers(1, self.buf)
def __init__(self, file):
self.name = file
wavefp = wave.open(file)
channels = wavefp.getnchannels()
bitrate = wavefp.getsampwidth() * 8
samplerate = wavefp.getframerate()
wavbuffer = wavefp.readframes(wavefp.getnframes())
self.duration = (len(wavbuffer) / float(samplerate)) / 2
self.length = len(wavbuffer)
formatmap = {
(1, 8): al.AL_FORMAT_MONO8,
(2, 8): al.AL_FORMAT_STEREO8,
(1, 16): al.AL_FORMAT_MONO16,
(2, 16): al.AL_FORMAT_STEREO16,
}
alformat = formatmap[(channels, bitrate)]
self.buffer = al.ALuint(0)
al.alGenBuffers(1, self.buffer)
al.alBufferData(self.buffer, alformat, wavbuffer, self.length,
samplerate)